Index: webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h |
diff --git a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h |
index 2deb4d243967779b7cfe790707311055aaf9ba0d..c18bac0d85b0aa5d16c45d98c01797716bdf9453 100644 |
--- a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h |
+++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h |
@@ -18,6 +18,7 @@ |
#include "webrtc/common_audio/lapped_transform.h" |
#include "webrtc/common_audio/channel_buffer.h" |
#include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h" |
+#include "webrtc/modules/audio_processing/vad/voice_activity_detector.h" |
namespace webrtc { |
@@ -28,28 +29,7 @@ namespace webrtc { |
// http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=6882788 |
class IntelligibilityEnhancer { |
public: |
- struct Config { |
- // TODO(bercic): the |decay_rate|, |analysis_rate| and |gain_limit| |
- // parameters should probably go away once fine tuning is done. |
- Config() |
- : sample_rate_hz(16000), |
- num_capture_channels(1), |
- num_render_channels(1), |
- decay_rate(0.9f), |
- analysis_rate(60), |
- gain_change_limit(0.1f), |
- rho(0.02f) {} |
- int sample_rate_hz; |
- size_t num_capture_channels; |
- size_t num_render_channels; |
- float decay_rate; |
- int analysis_rate; |
- float gain_change_limit; |
- float rho; |
- }; |
- |
- explicit IntelligibilityEnhancer(const Config& config); |
- IntelligibilityEnhancer(); // Initialize with default config. |
+ IntelligibilityEnhancer(int sample_rate_hz, size_t num_render_channels); |
// Sets the capture noise magnitude spectrum estimate. |
void SetCaptureNoiseEstimate(std::vector<float> noise); |
@@ -86,9 +66,6 @@ class IntelligibilityEnhancer { |
void ProcessClearBlock(const std::complex<float>* in_block, |
std::complex<float>* out_block); |
- // Computes and sets modified gains. |
- void AnalyzeClearBlock(); |
- |
// Bisection search for optimal |lambda|. |
void SolveForLambda(float power_target, float power_bot, float power_top); |
@@ -105,29 +82,25 @@ class IntelligibilityEnhancer { |
// Negative gains are set to 0. Stores the results in |sols|. |
void SolveForGainsGivenLambda(float lambda, size_t start_freq, float* sols); |
+ // Returns true if the audio is speech. |
+ bool IsSpeech(const float* audio); |
+ |
const size_t freqs_; // Num frequencies in frequency domain. |
- const size_t window_size_; // Window size in samples; also the block size. |
const size_t chunk_length_; // Chunk size in samples. |
const size_t bank_size_; // Num ERB filters. |
const int sample_rate_hz_; |
- const int erb_resolution_; |
- const size_t num_capture_channels_; |
const size_t num_render_channels_; |
- const int analysis_rate_; // Num blocks before gains recalculated. |
- const bool active_; // Whether render gains are being updated. |
- // TODO(ekm): Add logic for updating |active_|. |
- |
- intelligibility::PowerEstimator clear_power_; |
- std::vector<float> noise_power_; |
+ intelligibility::PowerEstimator<std::complex<float>> clear_power_estimator_; |
+ std::unique_ptr<intelligibility::PowerEstimator<float>> |
+ noise_power_estimator_; |
std::unique_ptr<float[]> filtered_clear_pow_; |
std::unique_ptr<float[]> filtered_noise_pow_; |
std::unique_ptr<float[]> center_freqs_; |
std::vector<std::vector<float>> capture_filter_bank_; |
std::vector<std::vector<float>> render_filter_bank_; |
size_t start_freq_; |
- std::unique_ptr<float[]> rho_; // Production and interpretation SNR. |
- // for each ERB band. |
+ |
std::unique_ptr<float[]> gains_eq_; // Pre-filter modified gains. |
intelligibility::GainApplier gain_applier_; |
@@ -135,11 +108,13 @@ class IntelligibilityEnhancer { |
// the original input array with modifications. |
ChannelBuffer<float> temp_render_out_buffer_; |
- std::unique_ptr<float[]> kbd_window_; |
TransformCallback render_callback_; |
std::unique_ptr<LappedTransform> render_mangler_; |
- int block_count_; |
- int analysis_step_; |
+ |
+ VoiceActivityDetector vad_; |
+ std::vector<int16_t> audio_s16_; |
+ size_t chunks_since_voice_; |
+ bool is_speech_; |
}; |
} // namespace webrtc |